Hi All,
I have a kamailio installed in my setup which is working properly in IPV4
configuration.
I want to perform IPV6 configuration, I don't have any idea how to
configure it,
I have tried few things by changing *kamailio.cfg* file but did not work.
I am sharing my *kamailio.cfg *file which is working fine in ipv4 Kindly
help me to configure for IPV6.
lets say my Ipv6 address is : 2402:3a80:d1a:f45a:fe44:2e54:ed2b:9bb2
Thanks,
Hi all
Having this issue with kamailio:Â Can't set module parameter
Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module <dispatcher>
Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 119, column 50: Can't set module parameter
Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module <dispatcher>
Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 120, column 50: Can't set module parameter
Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module <dispatcher>
Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 121, column 50: Can't set module parameter
here is full conf file: https://dpaste.de/bP3E
OS:
NAME="CentOS Linux"
VERSION="7 (Core)"Â
Software version:
kamailio 5.2.2 (x86_64/linux) 67f967
Thanks in advance,
tmbc
I think elaborating on the setup may help someone to provide inputs on this. Below is the setup.
Client 1 --> Kamailio proxy --> Backend SIP server
Client 2 --> Kamailio proxy --> Backend SIP server
Client 3 --> Kamailio proxy --> Backend SIP server
After I register all three clients, see only one TCP connection is being used between "Kamailio proxy - Backend SIP server".
Is there a way to achieve separate TCP connection to Backend SIP server from kamailio.
thanks,
raj
From: R, Rajkumar (Raj)
Sent: Monday, April 22, 2019 3:45 PM
To: sr-users(a)lists.kamailio.org
Subject: Config to control TCP connection reuse
Hi,
In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one.
Could anyone help how to achieve different TCP connections to same destination in kamailio?
thanks,
raj
Hi,
Recently, I started to use dmq_usrloc module. I successfully set it up.
Unfortunately, the documentation does not specify the behavior of
dmq_usrloc module with other Kamailio modules. For example, I see a bunch
of KDMQ messages are exchanged between nodes, but how the lookup() function
behaves with dmq_usrloc, or what is it's effect on a db_mode of 3!
Regards
I'm trying to send an SDP from a SIP client to Janus using JSON over HTTP.
The problem is that Janus is erroring on the SDP content with the following error:
JANUS HTTP: Get SDP for echo plugin: {#012Â Â "janus": "error",#012Â Â "error": {#012Â Â Â "code": 454,#012Â Â Â "reason": "JSON error: on line 9: control character 0xd near '\"v=0'"#012Â Â }#012} Result code 200
I tried using {s.escape.common} to escape any quotations, commas etc but it doesn't appear to replace the control characters with newlines.
Any suggestions on how best to remove these control characters so that Janus will accept the SDP over JSON?
Here is the relevant section of kamailio.cfg:
# HTTP: Request SDP for echo test plugin        sdp_get("$avp(sdp)");
        $var(res) = http_connect("janus", "/janus/$var(JANUS-ID)/$var(ECHO-ID)", "application/json", "{    \"janus\" : \"message\",    \"transaction\" : \"testtesttest99\",    \"body\" : {        \"audio\" : true    },    \"jsep\" : {        \"type\" : \"offer\",        \"sdp\" : \"$(avp(sdp){s.escape.common})\"    }}", "$avp(janus-pluginsession)");        xlog("L_INFO", "JANUS HTTP: Get SDP for echo plugin: $avp(janus-pluginsession) Result code $var(res)\n");
Greetings,
Is there a way to know in reply_route if a reply has direction downstream
or upstream regarding the first INVITE?
I know there are vars and methods for finding this out on requests, but i
can't find anything for replies.
Thanks in advance,
Best Regards,
Duarte Rocha
Hello,
I just noticed that the logged „start_time" as well as „end_time" of a canceled call is set to the „invite time“.
If done this way, it is harder to calculate the „ringing time“.
Is there an option to change this behaviour (i.e. to log the time of the received „bye“ instead)?
(If done so, the ringing time would always be the time between start and invite …)
Or is there a special reason for this behaviour?
My CDR settings are as followed:
modparam("acc", "cdr_enable", 1)
modparam("acc", "cdr_expired_dlg_enable", 1) # log unanswered / open dialogs
modparam("acc", "cdr_on_failed", 1) # log also not answered calls
modparam("acc", "cdr_extra", "inviteTime=$dlg_var(inviteTime)"
modparam("acc", "cdr_log_enable", 1) # activate write to syslog
modparam("acc", "cdr_facility", "LOG_LOCAL6") # syslog
modparam("acc", "cdr_start_on_confirmed", 1)
modparam("acc", "time_mode", 0) #syslog timestamp
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
modparam("acc", "detect_direction", 0)
modparam("acc", "log_flag", FLT_ACC) # needed for syslog
modparam("acc", "log_missed_flag", FLT_ACCMISSED) # need for logging missed calls in syslog
Regards
Jan-Hendrik
I'm using the following rtpengine_offer() to force the use of ICE relay and also replace o= and m=
  rtpengine_offer("replace-origin replace-session-connection ICE=force-relay RTP");
The SDP is being updated to include an ICE relay candidate, but the IP addresses in the o= and m= lines are not changing to the servers IP address (X.X.X.X).
I also tried using the media-address parameter but that didn't change the behaviour.
When I use rtpengine_answer() then it's not providing an ICE candidate nor changing o= or m=, which I imagine is expected.
What am I missing?
INVITE from the client (before rtpengine)------------------------------------------------------
INVITE sip:server.domain.com SIP/2.0Via: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rportFrom: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>Contact: <sip:username@154.20.1.8:37520;transport=tcp>Call-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Content-Length: 1506Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneSupported: 100rel
--7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp
v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=ptime:20a=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=rtpmap:114 AMR-WB/16000/1a=imageattr:114 octet-align=0a=fmtp:114 octet-align=0a=rtpmap:115 AMR-WB/16000/1a=imageattr:115 octet-align=1a=fmtp:115 octet-align=1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:101 0-16a=pcfg:1 t=1a=sendrecva=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audio
INVITE from the server (after rtpengine)------------------------------------------------------
SIP/2.0 200 OKVia: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rport=37520;received=2.221.73.230From: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>;tag=0ea7681ffd64b0ffcde1bff3393fd505.ae3aCall-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContact: <sip:X.X.X.X:443;transport=tcp>Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 1965
--7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp
v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=imageattr:114 octet-align=0a=imageattr:115 octet-align=1a=pcfg:1 t=1a=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audioa=rtpmap:114 AMR-WB/16000/1a=rtpmap:115 AMR-WB/16000/1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:114 octet-align=0a=fmtp:115 octet-align=1a=fmtp:101 0-16a=sendrecva=ptime:20a=ice-ufrag:kzgbFAnua=ice-pwd:hnEu9EwTrdIZnTWhYbxtmdugvra=candidate:dzkwmPcB28NPHfRA 1 UDP 16777215 X.X.X.X 30366 typ relay raddr X.X.X.X rport 30366a=candidate:dzkwmPcB28NPHfRA 2 UDP 16777214 X.X.X.X 30367 typ relay raddr X.X.X.X rport 30367a=sendrecva=ice-ufrag:uDE9wbvYa=ice-pwd:dx4CcIjqUNFfnRPJEF8GjatXdU
Hi,
In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one.
Could anyone help how to achieve different TCP connections to same destination in kamailio?
thanks,
raj
